Hallo Forum,
ich bekomme meinen QSC IPfonie extended -Trunk (Block 0-99) nicht an den Start:
Status: gelb “Es liegt ein Problem vor”
-> Eingehende Anrufe funktionieren nicht
Mein Sipgate Account funktioniert einwandfrei. Status grün.
-> Eingehende Anrufe funktionieren auf Voicebox oder Fax
Beide Provider habe ich über die Schnellanlage eingerichtet.
Folgende Parameter habe ich in der sip.conf hinzugefügt
http://community.pascom.net/showthread.php?31-Verbindungsproblem-mit-Sipgate
- externip <MeineExterneIP>
- localnet <meininternesnetz/maske>
Bisher leider ohne Erfolg.
Könnt Ihr helfen?
Danke,
pbxler
CLI:
NOTICE[3993]: chan_sip.c:5693 sip_reregister: – Re-registration for <MeinSipgateBenutzername>@sipgate.de
NOTICE[3993]: chan_sip.c:10483 handle_response_register: Outbound Registration: Expiry for sipgate.de is 600 sec (Scheduling reregistration in 585 s)
NOTICE[3993]: chan_sip.c:5693 sip_reregister: – Re-registration for <MeinQscBenutzername>@sip.qsc.de
sipNOTICE[3993]: chan_sip.c:10483 handle_response_register: Outbound Registration: Expiry for sip.qsc.de is 600 sec (Scheduling reregistration in 585 s)
sip debug:
md-3*CLI>
← SIP read from 217.10.79.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <MeineExterneIP>:5060;branch=z9hG4bK23b55637;rport=5060
From: “asterisk” <sip:asterisk@<MeineExterneIP>>;tag=as3c267f9c
To: <sip:sipgate.de>;tag=64c295986a77a1f756ad49f3e6513d0d.06cc
Call-ID: 6ad932955ebf13982a50f1622d23ada0@<MeineExterneIP>
CSeq: 102 OPTIONS
Accept: /
Accept-Encoding:
Accept-Language: en
Supported:
Content-Length: 0
— (11 headers 0 lines) —
Destroying call ‘6ad932955ebf13982a50f1622d23ada0@<MeineExterneIP>’
Jul 15 15:22:22 NOTICE[3993]: chan_sip.c:5693 sip_reregister: – Re-registration for <MeinSipgateBenutzername>@sipgate.de
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 217.10.79.9:5060:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP <MeineExterneIP>:5060;branch=z9hG4bK2396fb30;rport
From: <sip:<MeinSipgateBenutzername>@sipgate.de>;tag=as03c470d7
To: <sip:<MeinSipgateBenutzername>@sipgate.de>
Call-ID: 43c05ef46d4bc266728c585c6d8e9337@127.0.0.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 600
Contact: <sip:<MeinSipgateBenutzername>@<MeineExterneIP>>
Event: registration
Content-Length: 0
md-3*CLI>
← SIP read from 217.10.79.9:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <MeineExterneIP>:5060;received=<MeineExterneIP>;branch=z9hG4bK2396fb30;rport=5060
From: <sip:<MeinSipgateBenutzername>@sipgate.de>;tag=as03c470d7
To: <sip:<MeinSipgateBenutzername>@sipgate.de>;tag=fbf1d80521ea9f98078b6998e7669f9b.2e86
Call-ID: 43c05ef46d4bc266728c585c6d8e9337@127.0.0.1
CSeq: 102 REGISTER
WWW-Authenticate: Digest realm=“sipgate.de”, nonce=“5002c666da1464256561bad70cf1a985aa410bc4”
Content-Length: 0
— (8 headers 0 lines) —
Responding to challenge, registration to domain/host name sipgate.de
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 217.10.79.9:5060:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP <MeineExterneIP>:5060;branch=z9hG4bK5cac9d03;rport
From: <sip:<MeinSipgateBenutzername>@sipgate.de>;tag=as77d0ba89
To: <sip:<MeinSipgateBenutzername>@sipgate.de>
Call-ID: 43c05ef46d4bc266728c585c6d8e9337@127.0.0.1
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="<MeinSipgateBenutzername>", realm=“sipgate.de”, algorithm=MD5, uri=“sip:sipgate.de”, nonce=“5002c666da1464256561bad70cf1a985aa410bc4”, response=“d83ac6373934ba1cb0e19d679d866835”, opaque=""
Expires: 600
Contact: <sip:<MeinSipgateBenutzername>@<MeineExterneIP>>
Event: registration
Content-Length: 0
md-3*CLI>
← SIP read from 217.10.79.9:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <MeineExterneIP>:5060;received=<MeineExterneIP>;branch=z9hG4bK5cac9d03;rport=5060
From: <sip:<MeinSipgateBenutzername>@sipgate.de>;tag=as77d0ba89
To: <sip:<MeinSipgateBenutzername>@sipgate.de>;tag=fbf1d80521ea9f98078b6998e7669f9b.9ed1
Call-ID: 43c05ef46d4bc266728c585c6d8e9337@127.0.0.1
CSeq: 103 REGISTER
Contact: <sip:<MeinSipgateBenutzername>@<MeineExterneIP>>;expires=600
Content-Length: 0
— (8 headers 0 lines) —
Scheduling destruction of call ‘43c05ef46d4bc266728c585c6d8e9337@127.0.0.1’ in 32000 ms
Jul 15 15:22:22 NOTICE[3993]: chan_sip.c:10483 handle_response_register: Outbound Registration: Expiry for sipgate.de is 600 sec (Scheduling reregistration in 585 s)
Jul 15 15:22:22 NOTICE[3993]: chan_sip.c:5693 sip_reregister: – Re-registration for <MeinQscBenutzername>@sip.qsc.de
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 213.148.136.178:5060:
REGISTER sip:sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP <MeineExterneIP>:5060;branch=z9hG4bK528892f0;rport
From: <sip:<MeinQscBenutzername>@sip.qsc.de>;tag=as785635be
To: <sip:<MeinQscBenutzername>@sip.qsc.de>
Call-ID: 67e7059a34095f48037f88d54f6aa5e5@127.0.0.1
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 600
Contact: <sip:s@<MeineExterneIP>>
Event: registration
Content-Length: 0
md-3CLI>
← SIP read from 213.148.136.178:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <MeineExterneIP>:5060;received=<MeineExterneIP>;branch=z9hG4bK528892f0;rport
From: <sip:<MeinQscBenutzername>@sip.qsc.de>;tag=as785635be
To: <sip:<MeinQscBenutzername>@sip.qsc.de>;tag=SDuopv399-1f68ea9a5c07bd2ba275a961166f5dee.1424
Call-ID: 67e7059a34095f48037f88d54f6aa5e5@127.0.0.1
CSeq: 102 REGISTER
WWW-Authenticate: Digest realm=“qsc.de”, nonce=“UALJGVACx+0d/nx6eizGUqJ0mP0DYHNp”, qop=“auth”
Server: QSC SiP server node 03
Content-Length: 0
md-3CLI>
— (9 headers 0 lines) —
Responding to challenge, registration to domain/host name sip.qsc.de
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 213.148.136.178:5060:
REGISTER sip:sip.qsc.de SIP/2.0
Via: SIP/2.0/UDP <MeineExterneIP>:5060;branch=z9hG4bK70bfce76;rport
From: <sip:<MeinQscBenutzername>@sip.qsc.de>;tag=as71287df9
To: <sip:<MeinQscBenutzername>@sip.qsc.de>
Call-ID: 67e7059a34095f48037f88d54f6aa5e5@127.0.0.1
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="<MeinQscBenutzername>", realm=“qsc.de”, algorithm=MD5, uri=“sip:sip.qsc.de”, nonce=“UALJGVACx+0d/nx6eizGUqJ0mP0DYHNp”, response=“2748a89a749cc9f68db6c847d72107fc”, opaque="", qop=auth, cnonce=“29fed6e0”, nc=00000001
Expires: 600
Contact: <sip:s@<MeineExterneIP>>
Event: registration
Content-Length: 0
md-3*CLI>
← SIP read from 213.148.136.178:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <MeineExterneIP>:5060;received=<MeineExterneIP>;branch=z9hG4bK70bfce76;rport
From: <sip:<MeinQscBenutzername>@sip.qsc.de>;tag=as71287df9
To: <sip:<MeinQscBenutzername>@sip.qsc.de>;tag=SDuopv399-1f68ea9a5c07bd2ba275a961166f5dee.5212
Call-ID: 67e7059a34095f48037f88d54f6aa5e5@127.0.0.1
CSeq: 103 REGISTER
Contact: <sip:<MeinQscBenutzername>@<MeineExterneIP>>;expires=600
Server: QSC SiP server node 03
Content-Length: 0
Expires: 600
— (10 headers 0 lines) —
Scheduling destruction of call ‘67e7059a34095f48037f88d54f6aa5e5@127.0.0.1’ in 32000 ms
Jul 15 15:22:22 NOTICE[3993]: chan_sip.c:10483 handle_response_register: Outbound Registration: Expiry for sip.qsc.de is 600 sec (Scheduling reregistration in 585 s)
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager.conf.mdc’: Found
== Manager ‘phpasm’ logged on from 127.0.0.1
== Manager ‘phpasm’ logged off from 127.0.0.1
== Manager ‘phpasm’ logged off from 127.0.0.1
md-3*CLI>
sip debug Eingehender Anruf über QSC:
md-3*CLI>
← SIP read from 213.148.136.178:5060:
INVITE sip:<MeinQscBenutzername>@<MeineInterneIP>:5060 SIP/2.0
Via: SIP/2.0/UDP 213.148.136.178:5060;branch=z9hG4bKknf0sq203grgungt4001.1
Call-ID: SDd1k9a01-8a08868d6bd9eed11bbfdddb843bab30-l65h813
From: <sip:<MeineRufendeRufnummerl>@qsc.de;user=phone>;tag=SDd1k9a01-4411r859-CC-43
To: <sip:<MeineRufnummerMitDurchwahl>@qsc.de;user=phone>
CSeq: 1 INVITE
Max-Forwards: 63
P-Asserted-Identity: <sip:<MeineRufendeRufnummerl>@qsc.de;user=phone>
Contact: <sip:<MeineRufendeRufnummerl>@213.148.136.178:5060;transport=udp>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, PRACK, SUBSCRIBE, NOTIFY, UPDATE, MESSAGE, REFER
User-Agent: Huawei SoftX3000 V300R010
Supported: 100rel
P-Called-Party-ID: sip:<MeineRufnummerMitDurchwahl>@qsc.de
X-ORIGINAL-DDI-URI: sip:<MeineRufnummerMitDurchwahl>@qsc.de
X-ORIGINAL-DDI-USER: <MeineRufnummerMitDurchwahl>
X-CID: qah070q1eq1ln409589hrr78s7n5080l@SoftX3000
Content-Type: application/sdp
Content-Length: 321
v=0
o=HuaweiSoftX3000 16650002 16650002 IN IP4 213.148.136.178
s=Sip Call
c=IN IP4 213.148.136.178
t=0 0
m=audio 49192 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=fmtp:101 0-15
a=fmtp:18 annexb=yes
— (18 headers 14 lines) —
Using INVITE request as basis request - SDd1k9a01-8a08868d6bd9eed11bbfdddb843bab30-l65h813
Sending to 213.148.136.178 : 5060 (non-NAT)
Found no matching peer or user for ‘213.148.136.178:5060’
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 213.148.136.178:49192
Found description format PCMA
Found description format PCMU
Found description format G729
Found description format G723
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for <MeinQscBenutzername> in no-auth-in (domain <MeineInterneIP>)
list_route: hop: <sip:<MeineRufendeRufnummerl>@213.148.136.178:5060;transport=udp>
Transmitting (no NAT) to 213.148.136.178:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.148.136.178:5060;branch=z9hG4bKknf0sq203grgungt4001.1;received=213.148.136.178
From: <sip:<MeineRufendeRufnummerl>@qsc.de;user=phone>;tag=SDd1k9a01-4411r859-CC-43
To: <sip:<MeineRufnummerMitDurchwahl>@qsc.de;user=phone>
Call-ID: SDd1k9a01-8a08868d6bd9eed11bbfdddb843bab30-l65h813
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:<MeinQscBenutzername>@<MeineExterneIP>>
Content-Length: 0
-- Executing Macro("SIP/qsc.de-0838dc78", "mdc_emergency|<MeinQscBenutzername>") in new stack
-- Executing NoOp("SIP/qsc.de-0838dc78", "macro-mdc_emergency:: Exten: <MeinQscBenutzername>") in new stack
-- Executing Goto("SIP/qsc.de-0838dc78", "<MeinQscBenutzername>|1") in new stack
-- Goto (macro-mdc_emergency,<MeinQscBenutzername>,1)
-- Executing NoOp("SIP/qsc.de-0838dc78", "<MeinQscBenutzername> ist keine Notrufnummer") in new stack
-- Executing Hangup("SIP/qsc.de-0838dc78", "0") in new stack
== Spawn extension (no-auth-in, <MeinQscBenutzername>, 2) exited non-zero on ‘SIP/qsc.de-0838dc78’
Scheduling destruction of call ‘SDd1k9a01-8a08868d6bd9eed11bbfdddb843bab30-l65h813’ in 32000 ms
Reliably Transmitting (no NAT) to 213.148.136.178:5060:
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 213.148.136.178:5060;branch=z9hG4bKknf0sq203grgungt4001.1;received=213.148.136.178
From: <sip:<MeineRufendeRufnummerl>@qsc.de;user=phone>;tag=SDd1k9a01-4411r859-CC-43
To: <sip:<MeineRufnummerMitDurchwahl>@qsc.de;user=phone>;tag=as58ff8ca1
Call-ID: SDd1k9a01-8a08868d6bd9eed11bbfdddb843bab30-l65h813
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
md-3*CLI>
← SIP read from 213.148.136.178:5060:
ACK sip:<MeinQscBenutzername>@<MeineInterneIP>:5060 SIP/2.0
Via: SIP/2.0/UDP 213.148.136.178:5060;branch=z9hG4bKknf0sq203grgungt4001.1
CSeq: 1 ACK
Call-ID: SDd1k9a01-8a08868d6bd9eed11bbfdddb843bab30-l65h813
From: <sip:<MeineRufendeRufnummerl>@qsc.de;user=phone>;tag=SDd1k9a01-4411r859-CC-43
To: <sip:<MeineRufnummerMitDurchwahl>@qsc.de;user=phone>;tag=as58ff8ca1
Max-Forwards: 63
Content-Length: 0
— (8 headers 0 lines) —
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager.conf.mdc’: Found
== Manager ‘phpasm’ logged on from 127.0.0.1
== Manager ‘phpasm’ logged off from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager.conf.mdc’: Found
== Manager ‘phpasm’ logged on from 127.0.0.1
== Manager ‘phpasm’ logged off from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager.conf.mdc’: Found
== Manager ‘phpasm’ logged on from 127.0.0.1
== Manager ‘phpasm’ logged off from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager.conf.mdc’: Found
== Manager ‘phpasm’ logged on from 127.0.0.1
== Manager ‘phpasm’ logged off from 127.0.0.1
md-3CLI>
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager.conf.mdc’: Found
== Manager ‘phpasm’ logged on from 127.0.0.1
== Manager ‘phpasm’ logged off from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager.conf.mdc’: Found
== Manager ‘phpasm’ logged on from 127.0.0.1
== Manager ‘phpasm’ logged off from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager.conf.mdc’: Found
== Manager ‘phpasm’ logged on from 127.0.0.1
== Manager ‘phpasm’ logged off from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager.conf.mdc’: Found
== Manager ‘phpasm’ logged on from 127.0.0.1
== Manager ‘phpasm’ logged off from 127.0.0.1
md-3CLI>
← SIP read from 213.148.136.178:5060:
INVITE sip:<MeinQscBenutzername>@<MeineInterneIP>:5060 SIP/2.0
Via: SIP/2.0/UDP 213.148.136.178:5060;branch=z9hG4bK48o82620e89ghlkec661.1
Call-ID: SDtd9d801-b85fcc3d10424dc9f5d04ed40d099704-l65h813
From: <sip:<MeineRufendeRufnummerl>@qsc.de;user=phone>;tag=SDtd9d801-9ls2555a-CC-43
To: <sip:<MeineRufnummerMitDurchwahl>@qsc.de;user=phone>
CSeq: 1 INVITE
Max-Forwards: 63
P-Asserted-Identity: <sip:<MeineRufendeRufnummerl>@qsc.de;user=phone>
Contact: <sip:<MeineRufendeRufnummerl>@213.148.136.178:5060;transport=udp>
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, INFO, PRACK, SUBSCRIBE, NOTIFY, UPDATE, MESSAGE, REFER
User-Agent: Huawei SoftX3000 V300R010
Supported: 100rel
P-Called-Party-ID: sip:<MeineRufnummerMitDurchwahl>@qsc.de
X-ORIGINAL-DDI-URI: sip:<MeineRufnummerMitDurchwahl>@qsc.de
X-ORIGINAL-DDI-USER: <MeineRufnummerMitDurchwahl>
X-CID: 4n9re0r25he415rla17h58saeh420ql7@SoftX3000
Content-Type: application/sdp
Content-Length: 321
v=0
o=HuaweiSoftX3000 16650013 16650013 IN IP4 213.148.136.178
s=Sip Call
c=IN IP4 213.148.136.178
t=0 0
m=audio 55730 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=fmtp:101 0-15
a=fmtp:18 annexb=yes
— (18 headers 14 lines) —
Using INVITE request as basis request - SDtd9d801-b85fcc3d10424dc9f5d04ed40d099704-l65h813
Sending to 213.148.136.178 : 5060 (non-NAT)
Found no matching peer or user for ‘213.148.136.178:5060’
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 213.148.136.178:55730
Found description format PCMA
Found description format PCMU
Found description format G729
Found description format G723
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for <MeinQscBenutzername> in no-auth-in (domain <MeineInterneIP>)
list_route: hop: <sip:<MeineRufendeRufnummerl>@213.148.136.178:5060;transport=udp>
Transmitting (no NAT) to 213.148.136.178:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.148.136.178:5060;branch=z9hG4bK48o82620e89ghlkec661.1;received=213.148.136.178
From: <sip:<MeineRufendeRufnummerl>@qsc.de;user=phone>;tag=SDtd9d801-9ls2555a-CC-43
To: <sip:<MeineRufnummerMitDurchwahl>@qsc.de;user=phone>
Call-ID: SDtd9d801-b85fcc3d10424dc9f5d04ed40d099704-l65h813
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:<MeinQscBenutzername>@<MeineExterneIP>>
Content-Length: 0
-- Executing Macro("SIP/qsc.de-08401e30", "mdc_emergency|<MeinQscBenutzername>") in new stack
-- Executing NoOp("SIP/qsc.de-08401e30", "macro-mdc_emergency:: Exten: <MeinQscBenutzername>") in new stack
-- Executing Goto("SIP/qsc.de-08401e30", "<MeinQscBenutzername>|1") in new stack
-- Goto (macro-mdc_emergency,<MeinQscBenutzername>,1)
-- Executing NoOp("SIP/qsc.de-08401e30", "<MeinQscBenutzername> ist keine Notrufnummer") in new stack
-- Executing Hangup("SIP/qsc.de-08401e30", "0") in new stack
== Spawn extension (no-auth-in, <MeinQscBenutzername>, 2) exited non-zero on ‘SIP/qsc.de-08401e30’
Scheduling destruction of call ‘SDtd9d801-b85fcc3d10424dc9f5d04ed40d099704-l65h813’ in 32000 ms
Reliably Transmitting (no NAT) to 213.148.136.178:5060:
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 213.148.136.178:5060;branch=z9hG4bK48o82620e89ghlkec661.1;received=213.148.136.178
From: <sip:<MeineRufendeRufnummerl>@qsc.de;user=phone>;tag=SDtd9d801-9ls2555a-CC-43
To: <sip:<MeineRufnummerMitDurchwahl>@qsc.de;user=phone>;tag=as142bdd0a
Call-ID: SDtd9d801-b85fcc3d10424dc9f5d04ed40d099704-l65h813
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
md-3*CLI>
← SIP read from 213.148.136.178:5060:
ACK sip:<MeinQscBenutzername>@<MeineInterneIP>:5060 SIP/2.0
Via: SIP/2.0/UDP 213.148.136.178:5060;branch=z9hG4bK48o82620e89ghlkec661.1
CSeq: 1 ACK
Call-ID: SDtd9d801-b85fcc3d10424dc9f5d04ed40d099704-l65h813
From: <sip:<MeineRufendeRufnummerl>@qsc.de;user=phone>;tag=SDtd9d801-9ls2555a-CC-43
To: <sip:<MeineRufnummerMitDurchwahl>@qsc.de;user=phone>;tag=as142bdd0a
Max-Forwards: 63
Content-Length: 0
— (8 headers 0 lines) —
md-3*CLI>